[Asterisk-Users] Oh this is bad.... bindaddr and rtp traffic

[Alexander Lopez]

To retort, Digium has ever to my knowledge, stamped an 'Enterprise
Grade' mark on the product.  If you are worried about a single point of
failure you may want to replace your toaster.

Asterisk is missing a 'few features' no doubt about it, but it is open
source, it will be a welcome addition if you would like to add
multi-homing support in, might as well do media multi-homing with call
diversity. This will definably be a non-trivial re-architecture of the
core.

The 'missing a few features' way of thinking is what has made Asterisk
what it is today.

> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-
> bounces at lists.digium.com] On Behalf Of Douglas Garstang
> Sent: Tuesday, March 07, 2006 11:46 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Oh this is bad.... bindaddr and rtp
traffic
> 
> Pardon my candour, but for a product Digium calls 'enterprise grade'
it
> sure seems to be missing a few features.
> 
> -----Original Message-----
> From: Alexander Lopez [mailto:Alex.Lopez at OpSys.com]
> Sent: Tuesday, March 07, 2006 9:39 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Oh this is bad.... bindaddr and rtp
> traffic
> 
> 
> Asterisk does not like multiple interfaces in the way you are
configured.
> You can either:
> 
> A) use the bindaddr in the sip.conf to limit where the packsge come
and
> go.
> 
> B) use an outside traffic manager
> 
> Look up the archives, kpf explained why this would not work, as
asterisk
> can't do load balancing at this time
> 
> 
> -----Original Message-----
> From: "Robert Webb" <asterisk at ropeguru.com>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-
> users at lists.digium.com>
> Sent: 3/7/06 11:27 AM
> Subject: Re: [Asterisk-Users] Oh this is bad.... bindaddr and rtp
traffic
> 
> 
> On Tue, 7 Mar 2006 09:12:25 -0700
>   "Douglas Garstang" <dgarstang at oneeighty.com> wrote:
> > I have a configuration where RTP traffic is going out
> >interface pub0, and coming back into through pub1.
> > I have bindaddr=0.0.0.0 in sip.conf, and a netstat -an
> >shows:
> >
> > udp        0    788 0.0.0.0:5060            0.0.0.0:*
> >
> > which means that Asterisk is listening on all addresses
> >(on all interfaces?).
> >
> > Anyway, when the RTP traffic comes back in on interface
> >pub0, Asterisk does nothing with it. A 'rtp debug' shows
> >it's receiving the RTP packets, it just seems it does
> >nothing with them.
> >
> > Anyone seen this?
> >
> > Doug.
> >
> >
> 
> I thought all RTP was controlled through rtp.conf and only
> the SIP traffic was controlled through SIP.conf. I am not
> sure what settings, beside the RTP port range, you can out
> into the rtp.conf though.
> 
> Robert
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